aboutsummaryrefslogtreecommitdiffstats
path: root/fdk-aac/libAACenc/src/metadata_compressor.cpp
diff options
context:
space:
mode:
authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libAACenc/src/metadata_compressor.cpp
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
downloadODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz
ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2
ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libAACenc/src/metadata_compressor.cpp')
-rw-r--r--fdk-aac/libAACenc/src/metadata_compressor.cpp1579
1 files changed, 1579 insertions, 0 deletions
diff --git a/fdk-aac/libAACenc/src/metadata_compressor.cpp b/fdk-aac/libAACenc/src/metadata_compressor.cpp
new file mode 100644
index 0000000..bdac80a
--- /dev/null
+++ b/fdk-aac/libAACenc/src/metadata_compressor.cpp
@@ -0,0 +1,1579 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/**************************** AAC encoder library ******************************
+
+ Author(s): M. Neusinger
+
+ Description: Compressor for AAC Metadata Generator
+
+*******************************************************************************/
+
+#include "metadata_compressor.h"
+#include "channel_map.h"
+
+#define LOG2 0.69314718056f /* natural logarithm of 2 */
+#define ILOG2 1.442695041f /* 1/LOG2 */
+#define FIXP_ILOG2_DIV2 (FL2FXCONST_DBL(ILOG2 / 2))
+
+/*----------------- defines ----------------------*/
+
+#define MAX_DRC_CHANNELS (8) /*!< Max number of audio input channels. */
+#define DOWNMIX_SHIFT (3) /*!< Max 8 channel. */
+#define WEIGHTING_FILTER_SHIFT (2) /*!< Scaling used in weighting filter. */
+
+#define METADATA_INT_BITS 10
+#define METADATA_LINT_BITS 20
+#define METADATA_INT_SCALE (INT64(1) << (METADATA_INT_BITS))
+#define METADATA_FRACT_BITS (DFRACT_BITS - 1 - METADATA_INT_BITS)
+#define METADATA_FRACT_SCALE (INT64(1) << (METADATA_FRACT_BITS))
+
+/**
+ * Enum for channel assignment.
+ */
+enum { L = 0, R = 1, C = 2, LFE = 3, LS = 4, RS = 5, S = 6, LS2 = 7, RS2 = 8 };
+
+/*--------------- structure definitions --------------------*/
+
+/**
+ * Structure holds weighting filter filter states.
+ */
+struct WEIGHTING_STATES {
+ FIXP_DBL x1;
+ FIXP_DBL x2;
+ FIXP_DBL y1;
+ FIXP_DBL y2;
+};
+
+/**
+ * Dynamic Range Control compressor structure.
+ */
+struct DRC_COMP {
+ FIXP_DBL maxBoostThr[2]; /*!< Max boost threshold. */
+ FIXP_DBL boostThr[2]; /*!< Boost threshold. */
+ FIXP_DBL earlyCutThr[2]; /*!< Early cut threshold. */
+ FIXP_DBL cutThr[2]; /*!< Cut threshold. */
+ FIXP_DBL maxCutThr[2]; /*!< Max cut threshold. */
+
+ FIXP_DBL boostFac[2]; /*!< Precalculated factor for boost compression. */
+ FIXP_DBL
+ earlyCutFac[2]; /*!< Precalculated factor for early cut compression. */
+ FIXP_DBL cutFac[2]; /*!< Precalculated factor for cut compression. */
+
+ FIXP_DBL maxBoost[2]; /*!< Maximum boost. */
+ FIXP_DBL maxCut[2]; /*!< Maximum cut. */
+ FIXP_DBL maxEarlyCut[2]; /*!< Maximum early cut. */
+
+ FIXP_DBL fastAttack[2]; /*!< Fast attack coefficient. */
+ FIXP_DBL fastDecay[2]; /*!< Fast release coefficient. */
+ FIXP_DBL slowAttack[2]; /*!< Slow attack coefficient. */
+ FIXP_DBL slowDecay[2]; /*!< Slow release coefficient. */
+ UINT holdOff[2]; /*!< Hold time in blocks. */
+
+ FIXP_DBL attackThr[2]; /*!< Slow/fast attack threshold. */
+ FIXP_DBL decayThr[2]; /*!< Slow/fast release threshold. */
+
+ DRC_PROFILE profile[2]; /*!< DRC profile. */
+ INT blockLength; /*!< Block length in samples. */
+ UINT sampleRate; /*!< Sample rate. */
+ CHANNEL_MODE chanConfig; /*!< Channel configuration. */
+
+ UCHAR useWeighting; /*!< Use weighting filter. */
+
+ UINT channels; /*!< Number of channels. */
+ UINT fullChannels; /*!< Number of full range channels. */
+ INT channelIdx[9]; /*!< Offsets of interleaved channel samples (L, R, C, LFE,
+ Ls, Rs, S, Ls2, Rs2). */
+
+ FIXP_DBL smoothLevel[2]; /*!< level smoothing states */
+ FIXP_DBL smoothGain[2]; /*!< gain smoothing states */
+ UINT holdCnt[2]; /*!< hold counter */
+
+ FIXP_DBL limGain[2]; /*!< limiter gain */
+ FIXP_DBL limDecay; /*!< limiter decay (linear) */
+ FIXP_DBL prevPeak[2]; /*!< max peak of previous block (stereo/mono)*/
+
+ WEIGHTING_STATES
+ filter[MAX_DRC_CHANNELS]; /*!< array holds weighting filter states */
+};
+
+/*---------------- constants -----------------------*/
+
+/**
+ * Profile tables.
+ */
+static const FIXP_DBL tabMaxBoostThr[] = {
+ (FIXP_DBL)(-(43 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(53 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(55 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(65 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(50 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(40 << METADATA_FRACT_BITS))};
+static const FIXP_DBL tabBoostThr[] = {
+ (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(31 << METADATA_FRACT_BITS))};
+static const FIXP_DBL tabEarlyCutThr[] = {
+ (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(20 << METADATA_FRACT_BITS))};
+static const FIXP_DBL tabCutThr[] = {(FIXP_DBL)(-(16 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(11 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)),
+ (FIXP_DBL)(-(10 << METADATA_FRACT_BITS))};
+static const FIXP_DBL tabMaxCutThr[] = {
+ (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS)};
+static const FIXP_DBL tabBoostRatio[] = {
+ FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 5.f) - 1.f)), FL2FXCONST_DBL(((1.f / 5.f) - 1.f))};
+static const FIXP_DBL tabEarlyCutRatio[] = {
+ FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 1.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f))};
+static const FIXP_DBL tabCutRatio[] = {
+ FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)),
+ FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f))};
+static const FIXP_DBL tabMaxBoost[] = {(FIXP_DBL)(6 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(6 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(12 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(12 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS)};
+static const FIXP_DBL tabMaxCut[] = {(FIXP_DBL)(24 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(24 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(24 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(24 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(24 << METADATA_FRACT_BITS)};
+static const FIXP_DBL tabFastAttack[] = {
+ FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)};
+static const FIXP_DBL tabFastDecay[] = {
+ FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((200.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)};
+static const FIXP_DBL tabSlowAttack[] = {
+ FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)};
+static const FIXP_DBL tabSlowDecay[] = {
+ FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((10000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE),
+ FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)};
+
+static const INT tabHoldOff[] = {10, 10, 10, 10, 10, 0};
+
+static const FIXP_DBL tabAttackThr[] = {(FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(15 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(10 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(0 << METADATA_FRACT_BITS)};
+static const FIXP_DBL tabDecayThr[] = {(FIXP_DBL)(20 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(20 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(20 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(20 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(10 << METADATA_FRACT_BITS),
+ (FIXP_DBL)(0 << METADATA_FRACT_BITS)};
+
+/**
+ * Weighting filter coefficients (biquad bandpass).
+ */
+static const FIXP_DBL b0 = FL2FXCONST_DBL(0.53050662f); /* b1 = 0, b2 = -b0 */
+static const FIXP_DBL a1 = FL2FXCONST_DBL(-0.95237983f),
+ a2 = FL2FXCONST_DBL(-0.02248836f); /* a0 = 1 */
+
+/*------------- function definitions ----------------*/
+
+/**
+ * \brief Calculate scaling factor for denoted processing block.
+ *
+ * \param blockLength Length of processing block.
+ *
+ * \return shiftFactor
+ */
+static UINT getShiftFactor(const UINT length) {
+ UINT ldN;
+ for (ldN = 1; (((UINT)1) << ldN) < length; ldN++)
+ ;
+
+ return ldN;
+}
+
+/**
+ * \brief Sum up fixpoint values with best possible accuracy.
+ *
+ * \param value1 First input value.
+ * \param q1 Scaling factor of first input value.
+ * \param pValue2 Pointer to second input value, will be modified on
+ * return.
+ * \param pQ2 Pointer to second scaling factor, will be modified on
+ * return.
+ *
+ * \return void
+ */
+static void fixpAdd(const FIXP_DBL value1, const int q1,
+ FIXP_DBL* const pValue2, int* const pQ2) {
+ const int headroom1 = fNormz(fixp_abs(value1)) - 1;
+ const int headroom2 = fNormz(fixp_abs(*pValue2)) - 1;
+ int resultScale = fixMax(q1 - headroom1, (*pQ2) - headroom2);
+
+ if ((value1 != FL2FXCONST_DBL(0.f)) && (*pValue2 != FL2FXCONST_DBL(0.f))) {
+ resultScale++;
+ }
+
+ *pValue2 = scaleValue(value1, q1 - resultScale) +
+ scaleValue(*pValue2, (*pQ2) - resultScale);
+ *pQ2 = (*pValue2 != (FIXP_DBL)0) ? resultScale : DFRACT_BITS - 1;
+}
+
+/**
+ * \brief Function for converting time constant to filter coefficient.
+ *
+ * \param t Time constant.
+ * \param sampleRate Sampling rate in Hz.
+ * \param blockLength Length of processing block in samples per channel.
+ *
+ * \return result = 1.0 - exp(-1.0/((t) * (f)))
+ */
+static FIXP_DBL tc2Coeff(const FIXP_DBL t, const INT sampleRate,
+ const INT blockLength) {
+ FIXP_DBL sampleRateFract;
+ FIXP_DBL blockLengthFract;
+ FIXP_DBL f, product;
+ FIXP_DBL exponent, result;
+ INT e_res;
+
+ /* f = sampleRate/blockLength */
+ sampleRateFract =
+ (FIXP_DBL)(sampleRate << (DFRACT_BITS - 1 - METADATA_LINT_BITS));
+ blockLengthFract =
+ (FIXP_DBL)(blockLength << (DFRACT_BITS - 1 - METADATA_LINT_BITS));
+ f = fDivNorm(sampleRateFract, blockLengthFract, &e_res);
+ f = scaleValue(f, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */
+
+ /* product = t*f */
+ product = fMultNorm(t, f, &e_res);
+ product = scaleValue(
+ product, e_res + METADATA_INT_BITS); /* convert to METADATA_FRACT */
+
+ /* exponent = (-1.0/((t) * (f))) */
+ exponent = fDivNorm(METADATA_FRACT_SCALE, product, &e_res);
+ exponent = scaleValue(
+ exponent, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */
+
+ /* exponent * ld(e) */
+ exponent = fMult(exponent, FIXP_ILOG2_DIV2) << 1; /* e^(x) = 2^(x*ld(e)) */
+
+ /* exp(-1.0/((t) * (f))) */
+ result = f2Pow(-exponent, DFRACT_BITS - 1 - METADATA_FRACT_BITS, &e_res);
+
+ /* result = 1.0 - exp(-1.0/((t) * (f))) */
+ result = (FIXP_DBL)MAXVAL_DBL - scaleValue(result, e_res);
+
+ return result;
+}
+
+static void findPeakLevels(HDRC_COMP drcComp, const INT_PCM* const inSamples,
+ const FIXP_DBL clev, const FIXP_DBL slev,
+ const FIXP_DBL ext_leva, const FIXP_DBL ext_levb,
+ const FIXP_DBL lfe_lev, const FIXP_DBL dmxGain5,
+ const FIXP_DBL dmxGain2, FIXP_DBL peak[2]) {
+ int i, c;
+ FIXP_DBL tmp = FL2FXCONST_DBL(0.f);
+ INT_PCM maxSample = 0;
+
+ /* find peak level */
+ peak[0] = peak[1] = FL2FXCONST_DBL(0.f);
+ for (i = 0; i < drcComp->blockLength; i++) {
+ const INT_PCM* pSamples = &inSamples[i * drcComp->channels];
+
+ /* single channels */
+ for (c = 0; c < (int)drcComp->channels; c++) {
+ maxSample = fMax(maxSample, (INT_PCM)fAbs(pSamples[c]));
+ }
+ }
+ peak[0] = fixMax(peak[0], FX_PCM2FX_DBL(maxSample) >> DOWNMIX_SHIFT);
+
+ /* 7.1/6.1 to 5.1 downmixes */
+ if (drcComp->fullChannels > 5) {
+ for (i = 0; i < drcComp->blockLength; i++) {
+ const INT_PCM* pSamples = &inSamples[i * drcComp->channels];
+
+ /* channel 1 (L, Ls,...) */
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_6_1:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >>
+ (DOWNMIX_SHIFT - 1); /* Cs */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lrs / Lss */
+ break;
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ (DOWNMIX_SHIFT - 1); /* L */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lvh */
+ break;
+ default:
+ break;
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+
+ /* channel 2 (R, Rs,...) */
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_6_1:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >>
+ (DOWNMIX_SHIFT - 1); /* Cs */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rrs / Rss */
+ break;
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ (DOWNMIX_SHIFT - 1); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rvh */
+ break;
+ default:
+ break;
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+
+ /* channel 3 (C) */
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ DOWNMIX_SHIFT); /* C */
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc */
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc */
+ break;
+ default:
+ break;
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+
+ } /* for (blocklength) */
+
+ /* take downmix gain into accout */
+ peak[0] = fMult(dmxGain5, peak[0])
+ << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+
+ /* 7.1 / 5.1 to stereo downmixes */
+ if (drcComp->fullChannels > 2) {
+ /* Lt/Rt downmix */
+ for (i = 0; i < drcComp->blockLength; i++) {
+ const INT_PCM* pSamples = &inSamples[i * drcComp->channels];
+
+ /* Lt */
+ tmp = FL2FXCONST_DBL(0.f);
+ if (drcComp->channelIdx[LS] >= 0)
+ tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ if (drcComp->channelIdx[LS2] >= 0)
+ tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls2 */
+ if (drcComp->channelIdx[RS] >= 0)
+ tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ if (drcComp->channelIdx[RS2] >= 0)
+ tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs2 */
+ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0))
+ tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */
+ if (drcComp->channelIdx[S] >= 0)
+ tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >>
+ (DOWNMIX_SHIFT - 1); /* S */
+ if (drcComp->channelIdx[C] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >>
+ DOWNMIX_SHIFT); /* L */
+
+ /* apply scaling of downmix gains */
+ /* only for positive values only, as legacy decoders might not know this
+ * parameter */
+ if (dmxGain2 > FL2FXCONST_DBL(0.f)) {
+ if (drcComp->fullChannels > 5) {
+ tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+
+ /* Rt */
+ tmp = FL2FXCONST_DBL(0.f);
+ if (drcComp->channelIdx[LS] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ if (drcComp->channelIdx[LS2] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls2 */
+ if (drcComp->channelIdx[RS] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ if (drcComp->channelIdx[RS2] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs2 */
+ if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0))
+ tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */
+ if (drcComp->channelIdx[S] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >>
+ (DOWNMIX_SHIFT - 1); /* S */
+ if (drcComp->channelIdx[C] >= 0)
+ tmp += fMultDiv2(FL2FXCONST_DBL(0.707f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >>
+ DOWNMIX_SHIFT); /* R */
+
+ /* apply scaling of downmix gains */
+ /* only for positive values only, as legacy decoders might not know this
+ * parameter */
+ if (dmxGain2 > FL2FXCONST_DBL(0.f)) {
+ if (drcComp->fullChannels > 5) {
+ tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+ }
+
+ /* Lo/Ro downmix */
+ for (i = 0; i < drcComp->blockLength; i++) {
+ const INT_PCM* pSamples = &inSamples[i * drcComp->channels];
+
+ /* Lo */
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_6_1:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >>
+ (DOWNMIX_SHIFT - 1); /* Cs */
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc */
+ tmp += fMultDiv2(fMult(ext_leva, clev),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc - second path*/
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ (DOWNMIX_SHIFT - 1); /* L */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lvh */
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ default:
+ if (drcComp->channelIdx[LS] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ if (drcComp->channelIdx[LS2] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls2 */
+ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0))
+ tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */
+ if (drcComp->channelIdx[S] >= 0)
+ tmp +=
+ fMultDiv2(slev,
+ fMult(FL2FXCONST_DBL(0.7f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >>
+ (DOWNMIX_SHIFT - 1); /* S */
+ if (drcComp->channelIdx[C] >= 0)
+ tmp +=
+ fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ if (drcComp->channelIdx[3] >= 0)
+ tmp += fMultDiv2(lfe_lev,
+ (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >>
+ DOWNMIX_SHIFT); /* L */
+ break;
+ }
+
+ /* apply scaling of downmix gains */
+ /* only for positive values only, as legacy decoders might not know this
+ * parameter */
+ if (dmxGain2 > FL2FXCONST_DBL(0.f)) {
+ if (drcComp->fullChannels > 5) {
+ tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+
+ /* Ro */
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_6_1:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >>
+ (DOWNMIX_SHIFT - 1); /* Cs */
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc */
+ tmp += fMultDiv2(fMult(ext_leva, clev),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc - second path*/
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ (DOWNMIX_SHIFT - 1); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rvh */
+ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp +=
+ fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ default:
+ if (drcComp->channelIdx[RS] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ if (drcComp->channelIdx[RS2] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs2 */
+ if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0))
+ tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */
+ if (drcComp->channelIdx[S] >= 0)
+ tmp +=
+ fMultDiv2(slev,
+ fMult(FL2FXCONST_DBL(0.7f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >>
+ (DOWNMIX_SHIFT - 1); /* S */
+ if (drcComp->channelIdx[C] >= 0)
+ tmp +=
+ fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ if (drcComp->channelIdx[3] >= 0)
+ tmp += fMultDiv2(lfe_lev,
+ (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >>
+ DOWNMIX_SHIFT); /* R */
+ }
+
+ /* apply scaling of downmix gains */
+ /* only for positive values only, as legacy decoders might not know this
+ * parameter */
+ if (dmxGain2 > FL2FXCONST_DBL(0.f)) {
+ if (drcComp->fullChannels > 5) {
+ tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ peak[0] = fixMax(peak[0], fixp_abs(tmp));
+ }
+ }
+
+ peak[1] = fixMax(peak[0], peak[1]);
+
+ /* Mono Downmix - for comp_val only */
+ if (drcComp->fullChannels > 1) {
+ for (i = 0; i < drcComp->blockLength; i++) {
+ const INT_PCM* pSamples = &inSamples[i * drcComp->channels];
+
+ tmp = FL2FXCONST_DBL(0.f);
+ switch (drcComp->chanConfig) {
+ case MODE_6_1:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMult(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >>
+ (DOWNMIX_SHIFT - 1); /* Cs */
+ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(fMult(slev, ext_leva),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/
+ tmp += fMultDiv2(fMult(slev, ext_levb),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/
+ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_1_2_2_2_1:
+ case MODE_7_1_FRONT_CENTER:
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ DOWNMIX_SHIFT); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc */
+ tmp += fMultDiv2(fMult(ext_leva, clev),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lc - second path*/
+ tmp += fMultDiv2(fMult(ext_leva, clev),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rc - second path*/
+ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >>
+ (DOWNMIX_SHIFT - 1); /* L */
+ tmp +=
+ fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >>
+ (DOWNMIX_SHIFT - 1); /* R */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >>
+ (DOWNMIX_SHIFT - 1); /* Lvh */
+ tmp +=
+ fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rvh */
+ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >>
+ (DOWNMIX_SHIFT - 1); /* C */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ break;
+ default:
+ if (drcComp->channelIdx[LS] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls */
+ if (drcComp->channelIdx[LS2] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Ls2 */
+ if (drcComp->channelIdx[RS] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs */
+ if (drcComp->channelIdx[RS2] >= 0)
+ tmp +=
+ fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >>
+ (DOWNMIX_SHIFT - 1); /* Rs2 */
+ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0))
+ tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */
+ /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/ /* 7.1ch */
+ if (drcComp->channelIdx[S] >= 0)
+ tmp +=
+ fMultDiv2(slev,
+ fMult(FL2FXCONST_DBL(0.7f),
+ (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >>
+ (DOWNMIX_SHIFT - 1); /* S */
+ if (drcComp->channelIdx[C] >= 0)
+ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >>
+ (DOWNMIX_SHIFT - 1); /* C (2*clev) */
+ if (drcComp->channelIdx[3] >= 0)
+ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >>
+ (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >>
+ DOWNMIX_SHIFT); /* L */
+ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >>
+ DOWNMIX_SHIFT); /* R */
+ }
+
+ /* apply scaling of downmix gains */
+ /* only for positive values only, as legacy decoders might not know this
+ * parameter */
+ if (dmxGain2 > FL2FXCONST_DBL(0.f)) {
+ if (drcComp->fullChannels > 5) {
+ tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS);
+ }
+ peak[1] = fixMax(peak[1], fixp_abs(tmp));
+ }
+ }
+}
+
+INT FDK_DRC_Generator_Open(HDRC_COMP* phDrcComp) {
+ INT err = 0;
+ HDRC_COMP hDcComp = NULL;
+
+ if (phDrcComp == NULL) {
+ err = -1;
+ goto bail;
+ }
+
+ /* allocate memory */
+ hDcComp = (HDRC_COMP)FDKcalloc(1, sizeof(DRC_COMP));
+
+ if (hDcComp == NULL) {
+ err = -1;
+ goto bail;
+ }
+
+ FDKmemclear(hDcComp, sizeof(DRC_COMP));
+
+ /* Return drc compressor instance */
+ *phDrcComp = hDcComp;
+ return err;
+bail:
+ FDK_DRC_Generator_Close(&hDcComp);
+ return err;
+}
+
+INT FDK_DRC_Generator_Close(HDRC_COMP* phDrcComp) {
+ if (phDrcComp == NULL) {
+ return -1;
+ }
+ if (*phDrcComp != NULL) {
+ FDKfree(*phDrcComp);
+ *phDrcComp = NULL;
+ }
+ return 0;
+}
+
+INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp,
+ const DRC_PROFILE profileLine,
+ const DRC_PROFILE profileRF,
+ const INT blockLength, const UINT sampleRate,
+ const CHANNEL_MODE channelMode,
+ const CHANNEL_ORDER channelOrder,
+ const UCHAR useWeighting) {
+ int i;
+ CHANNEL_MAPPING channelMapping;
+
+ drcComp->limDecay =
+ FL2FXCONST_DBL(((0.006f / 256) * blockLength) / METADATA_INT_SCALE);
+
+ /* Save parameters. */
+ drcComp->blockLength = blockLength;
+ drcComp->sampleRate = sampleRate;
+ drcComp->chanConfig = channelMode;
+ drcComp->useWeighting = useWeighting;
+
+ if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF) !=
+ 0) { /* expects initialized blockLength and sampleRate */
+ return (-1);
+ }
+
+ /* Set number of channels and channel offsets. */
+ if (FDKaacEnc_InitChannelMapping(channelMode, channelOrder,
+ &channelMapping) != AAC_ENC_OK) {
+ return (-2);
+ }
+
+ for (i = 0; i < 9; i++) drcComp->channelIdx[i] = -1;
+
+ switch (channelMode) {
+ case MODE_1: /* mono */
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0];
+ break;
+ case MODE_2: /* stereo */
+ drcComp->channelIdx[L] = channelMapping.elInfo[0].ChannelIndex[0];
+ drcComp->channelIdx[R] = channelMapping.elInfo[0].ChannelIndex[1];
+ break;
+ case MODE_1_2: /* 3ch */
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0];
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1];
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0];
+ break;
+ case MODE_1_2_1: /* 4ch */
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0];
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1];
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0];
+ drcComp->channelIdx[S] = channelMapping.elInfo[2].ChannelIndex[0];
+ break;
+ case MODE_1_2_2: /* 5ch */
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0];
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1];
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0];
+ drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0];
+ drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1];
+ break;
+ case MODE_1_2_2_1: /* 5.1 ch */
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0];
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1];
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0];
+ drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0];
+ drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0];
+ drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1];
+ break;
+ case MODE_1_2_2_2_1: /* 7.1 ch */
+ case MODE_7_1_FRONT_CENTER:
+ drcComp->channelIdx[L] = channelMapping.elInfo[2].ChannelIndex[0]; /* l */
+ drcComp->channelIdx[R] = channelMapping.elInfo[2].ChannelIndex[1]; /* r */
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+ drcComp->channelIdx[LFE] =
+ channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */
+ drcComp->channelIdx[LS] =
+ channelMapping.elInfo[3].ChannelIndex[0]; /* ls */
+ drcComp->channelIdx[RS] =
+ channelMapping.elInfo[3].ChannelIndex[1]; /* rs */
+ drcComp->channelIdx[LS2] =
+ channelMapping.elInfo[1].ChannelIndex[0]; /* lc */
+ drcComp->channelIdx[RS2] =
+ channelMapping.elInfo[1].ChannelIndex[1]; /* rc */
+ break;
+ case MODE_7_1_BACK:
+ case MODE_7_1_REAR_SURROUND:
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+ drcComp->channelIdx[LFE] =
+ channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */
+ drcComp->channelIdx[LS] =
+ channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */
+ drcComp->channelIdx[RS] =
+ channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */
+ drcComp->channelIdx[LS2] =
+ channelMapping.elInfo[2].ChannelIndex[0]; /* ls */
+ drcComp->channelIdx[RS2] =
+ channelMapping.elInfo[2].ChannelIndex[1]; /* rs */
+ break;
+ case MODE_6_1:
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+ drcComp->channelIdx[LFE] =
+ channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */
+ drcComp->channelIdx[LS] =
+ channelMapping.elInfo[2].ChannelIndex[0]; /* ls */
+ drcComp->channelIdx[RS] =
+ channelMapping.elInfo[2].ChannelIndex[1]; /* rs */
+ drcComp->channelIdx[S] = channelMapping.elInfo[3].ChannelIndex[0]; /* s */
+ break;
+ case MODE_7_1_TOP_FRONT:
+ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */
+ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */
+ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+ drcComp->channelIdx[LFE] =
+ channelMapping.elInfo[3].ChannelIndex[0]; /* lfe */
+ drcComp->channelIdx[LS] =
+ channelMapping.elInfo[2].ChannelIndex[0]; /* ls */
+ drcComp->channelIdx[RS] =
+ channelMapping.elInfo[2].ChannelIndex[1]; /* rs */
+ drcComp->channelIdx[LS2] =
+ channelMapping.elInfo[4].ChannelIndex[0]; /* lvh2 */
+ drcComp->channelIdx[RS2] =
+ channelMapping.elInfo[4].ChannelIndex[1]; /* rvh2 */
+ break;
+ default:
+ return (-1);
+ }
+
+ drcComp->fullChannels = channelMapping.nChannelsEff;
+ drcComp->channels = channelMapping.nChannels;
+
+ /* Init states. */
+ drcComp->smoothLevel[0] = drcComp->smoothLevel[1] =
+ (FIXP_DBL)(-(135 << METADATA_FRACT_BITS));
+
+ FDKmemclear(drcComp->smoothGain, sizeof(drcComp->smoothGain));
+ FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt));
+ FDKmemclear(drcComp->limGain, sizeof(drcComp->limGain));
+ FDKmemclear(drcComp->prevPeak, sizeof(drcComp->prevPeak));
+ FDKmemclear(drcComp->filter, sizeof(drcComp->filter));
+
+ return (0);
+}
+
+INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp,
+ const DRC_PROFILE profileLine,
+ const DRC_PROFILE profileRF) {
+ int profileIdx, i;
+
+ drcComp->profile[0] = profileLine;
+ drcComp->profile[1] = profileRF;
+
+ for (i = 0; i < 2; i++) {
+ /* get profile index */
+ switch (drcComp->profile[i]) {
+ case DRC_NONE:
+ case DRC_NOT_PRESENT:
+ case DRC_FILMSTANDARD:
+ profileIdx = 0;
+ break;
+ case DRC_FILMLIGHT:
+ profileIdx = 1;
+ break;
+ case DRC_MUSICSTANDARD:
+ profileIdx = 2;
+ break;
+ case DRC_MUSICLIGHT:
+ profileIdx = 3;
+ break;
+ case DRC_SPEECH:
+ profileIdx = 4;
+ break;
+ case DRC_DELAY_TEST:
+ profileIdx = 5;
+ break;
+ default:
+ return (-1);
+ }
+
+ /* get parameters for selected profile */
+ if (profileIdx >= 0) {
+ drcComp->maxBoostThr[i] = tabMaxBoostThr[profileIdx];
+ drcComp->boostThr[i] = tabBoostThr[profileIdx];
+ drcComp->earlyCutThr[i] = tabEarlyCutThr[profileIdx];
+ drcComp->cutThr[i] = tabCutThr[profileIdx];
+ drcComp->maxCutThr[i] = tabMaxCutThr[profileIdx];
+
+ drcComp->boostFac[i] = tabBoostRatio[profileIdx];
+ drcComp->earlyCutFac[i] = tabEarlyCutRatio[profileIdx];
+ drcComp->cutFac[i] = tabCutRatio[profileIdx];
+
+ drcComp->maxBoost[i] = tabMaxBoost[profileIdx];
+ drcComp->maxCut[i] = tabMaxCut[profileIdx];
+ drcComp->maxEarlyCut[i] =
+ -fMult((drcComp->cutThr[i] - drcComp->earlyCutThr[i]),
+ drcComp->earlyCutFac[i]); /* no scaling after mult needed,
+ earlyCutFac is in FIXP_DBL */
+
+ drcComp->fastAttack[i] = tc2Coeff(
+ tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength);
+ drcComp->fastDecay[i] = tc2Coeff(
+ tabFastDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength);
+ drcComp->slowAttack[i] = tc2Coeff(
+ tabSlowAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength);
+ drcComp->slowDecay[i] = tc2Coeff(
+ tabSlowDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength);
+ drcComp->holdOff[i] = tabHoldOff[profileIdx] * 256 / drcComp->blockLength;
+
+ drcComp->attackThr[i] = tabAttackThr[profileIdx];
+ drcComp->decayThr[i] = tabDecayThr[profileIdx];
+ }
+
+ drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f);
+ }
+ return (0);
+}
+
+INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM* const inSamples,
+ const UINT inSamplesBufSize, const INT dialnorm,
+ const INT drc_TargetRefLevel,
+ const INT comp_TargetRefLevel, const FIXP_DBL clev,
+ const FIXP_DBL slev, const FIXP_DBL ext_leva,
+ const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev,
+ const INT dmxGain5, const INT dmxGain2,
+ INT* const pDynrng, INT* const pCompr) {
+ int i, c;
+ FIXP_DBL peak[2];
+
+ /**************************************************************************
+ * compressor
+ **************************************************************************/
+ if ((drcComp->profile[0] != DRC_NONE) || (drcComp->profile[1] != DRC_NONE)) {
+ /* Calc loudness level */
+ FIXP_DBL level_b = FL2FXCONST_DBL(0.f);
+ int level_e = DFRACT_BITS - 1;
+
+ /* Increase energy time resolution with shorter processing blocks. 16 is an
+ * empiric value. */
+ const int granuleLength = fixMin(16, drcComp->blockLength);
+
+ if (drcComp->useWeighting) {
+ FIXP_DBL x1, x2, y, y1, y2;
+ /* sum of filter coefficients about 2.5 -> squared value is 6.25
+ WEIGHTING_FILTER_SHIFT is 2 -> scaling about 16, therefore reduce
+ granuleShift by 1.
+ */
+ const int granuleShift = getShiftFactor(granuleLength) - 1;
+
+ for (c = 0; c < (int)drcComp->channels; c++) {
+ const INT_PCM* pSamples = inSamples + c * inSamplesBufSize;
+
+ if (c == drcComp->channelIdx[LFE]) {
+ continue; /* skip LFE */
+ }
+
+ /* get filter states */
+ x1 = drcComp->filter[c].x1;
+ x2 = drcComp->filter[c].x2;
+ y1 = drcComp->filter[c].y1;
+ y2 = drcComp->filter[c].y2;
+
+ i = 0;
+
+ do {
+ int offset = i;
+ FIXP_DBL accu = FL2FXCONST_DBL(0.f);
+
+ for (i = offset;
+ i < fixMin(offset + granuleLength, drcComp->blockLength); i++) {
+ /* apply weighting filter */
+ FIXP_DBL x =
+ FX_PCM2FX_DBL((FIXP_PCM)pSamples[i]) >> WEIGHTING_FILTER_SHIFT;
+
+ /* y = b0 * (x - x2) - a1 * y1 - a2 * y2; */
+ y = fMult(b0, x - x2) - fMult(a1, y1) - fMult(a2, y2);
+
+ x2 = x1;
+ x1 = x;
+ y2 = y1;
+ y1 = y;
+
+ accu += fPow2Div2(y) >> (granuleShift - 1); /* partial energy */
+ } /* i */
+
+ fixpAdd(accu, granuleShift + 2 * WEIGHTING_FILTER_SHIFT, &level_b,
+ &level_e); /* sup up partial energies */
+
+ } while (i < drcComp->blockLength);
+
+ /* save filter states */
+ drcComp->filter[c].x1 = x1;
+ drcComp->filter[c].x2 = x2;
+ drcComp->filter[c].y1 = y1;
+ drcComp->filter[c].y2 = y2;
+ } /* c */
+ } /* weighting */
+ else {
+ const int granuleShift = getShiftFactor(granuleLength);
+
+ for (c = 0; c < (int)drcComp->channels; c++) {
+ const INT_PCM* pSamples = inSamples + c * inSamplesBufSize;
+
+ if ((int)c == drcComp->channelIdx[LFE]) {
+ continue; /* skip LFE */
+ }
+
+ i = 0;
+
+ do {
+ int offset = i;
+ FIXP_DBL accu = FL2FXCONST_DBL(0.f);
+
+ for (i = offset;
+ i < fixMin(offset + granuleLength, drcComp->blockLength); i++) {
+ /* partial energy */
+ accu += fPow2Div2((FIXP_PCM)pSamples[i]) >> (granuleShift - 1);
+ } /* i */
+
+ fixpAdd(accu, granuleShift, &level_b,
+ &level_e); /* sup up partial energies */
+
+ } while (i < drcComp->blockLength);
+ }
+ } /* weighting */
+
+ /*
+ * Convert to dBFS, apply dialnorm
+ */
+ /* level scaling */
+
+ /* descaled level in ld64 representation */
+ FIXP_DBL ldLevel =
+ CalcLdData(level_b) +
+ (FIXP_DBL)((level_e - 12) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)) -
+ CalcLdData((FIXP_DBL)(drcComp->blockLength << (DFRACT_BITS - 1 - 12)));
+
+ /* if (level < 1e-10) level = 1e-10f; */
+ ldLevel =
+ fMax(ldLevel, FL2FXCONST_DBL(-0.51905126482615036685473741085772f));
+
+ /* level = 10 * log(level)/log(10) + 3;
+ * = 10*log(2)/log(10) * ld(level) + 3;
+ * = 10 * 0.30102999566398119521373889472449 * ld(level) + 3
+ * = 10 * (0.30102999566398119521373889472449 * ld(level) + 0.3)
+ * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64)
+ * * 64
+ *
+ * additional scaling with METADATA_FRACT_BITS:
+ * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64)
+ * * 64 * 2^(METADATA_FRACT_BITS) = 10 * (0.30102999566398119521373889472449
+ * * ld64(level) + 0.3/64) * 2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) =
+ * 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * (
+ * 0.30102999566398119521373889472449 * ld64(level) + 0.3/64 )
+ * */
+ FIXP_DBL level = fMult(
+ (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)),
+ fMult(FL2FXCONST_DBL(0.30102999566398119521373889472449f), ldLevel) +
+ (FIXP_DBL)(FL2FXCONST_DBL(0.3f) >> LD_DATA_SHIFT));
+
+ /* level -= dialnorm + 31 */ /* this is fixed to Dolby-ReferenceLevel as
+ compressor profiles are defined relative to
+ this */
+ level -= ((FIXP_DBL)(dialnorm << (METADATA_FRACT_BITS - 16)) +
+ (FIXP_DBL)(31 << METADATA_FRACT_BITS));
+
+ for (i = 0; i < 2; i++) {
+ if (drcComp->profile[i] == DRC_NONE) {
+ /* no compression */
+ drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f);
+ } else {
+ FIXP_DBL gain, alpha, lvl2smthlvl;
+
+ /* calc static gain */
+ if (level <= drcComp->maxBoostThr[i]) {
+ /* max boost */
+ gain = drcComp->maxBoost[i];
+ } else if (level < drcComp->boostThr[i]) {
+ /* boost range */
+ gain = fMult((level - drcComp->boostThr[i]), drcComp->boostFac[i]);
+ } else if (level <= drcComp->earlyCutThr[i]) {
+ /* null band */
+ gain = FL2FXCONST_DBL(0.f);
+ } else if (level <= drcComp->cutThr[i]) {
+ /* early cut range */
+ gain =
+ fMult((level - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]);
+ } else if (level < drcComp->maxCutThr[i]) {
+ /* cut range */
+ gain = fMult((level - drcComp->cutThr[i]), drcComp->cutFac[i]) -
+ drcComp->maxEarlyCut[i];
+ } else {
+ /* max cut */
+ gain = -drcComp->maxCut[i];
+ }
+
+ /* choose time constant */
+ lvl2smthlvl = level - drcComp->smoothLevel[i];
+ if (gain < drcComp->smoothGain[i]) {
+ /* attack */
+ if (lvl2smthlvl > drcComp->attackThr[i]) {
+ /* fast attack */
+ alpha = drcComp->fastAttack[i];
+ } else {
+ /* slow attack */
+ alpha = drcComp->slowAttack[i];
+ }
+ } else {
+ /* release */
+ if (lvl2smthlvl < -drcComp->decayThr[i]) {
+ /* fast release */
+ alpha = drcComp->fastDecay[i];
+ } else {
+ /* slow release */
+ alpha = drcComp->slowDecay[i];
+ }
+ }
+
+ /* smooth gain & level */
+ if ((gain < drcComp->smoothGain[i]) ||
+ (drcComp->holdCnt[i] ==
+ 0)) { /* hold gain unless we have an attack or hold
+ period is over */
+ FIXP_DBL accu;
+
+ /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] +
+ * alpha * level; */
+ accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothLevel[i]);
+ accu += fMult(alpha, level);
+ drcComp->smoothLevel[i] = accu;
+
+ /* drcComp->smoothGain[i] = (1-alpha) * drcComp->smoothGain[i] +
+ * alpha * gain; */
+ accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothGain[i]);
+ accu += fMult(alpha, gain);
+ drcComp->smoothGain[i] = accu;
+ }
+
+ /* hold counter */
+ if (drcComp->holdCnt[i]) {
+ drcComp->holdCnt[i]--;
+ }
+ if (gain < drcComp->smoothGain[i]) {
+ drcComp->holdCnt[i] = drcComp->holdOff[i];
+ }
+ } /* profile != DRC_NONE */
+ } /* for i=1..2 */
+ } else {
+ /* no compression */
+ drcComp->smoothGain[0] = FL2FXCONST_DBL(0.f);
+ drcComp->smoothGain[1] = FL2FXCONST_DBL(0.f);
+ }
+
+ /**************************************************************************
+ * limiter
+ **************************************************************************/
+
+ findPeakLevels(drcComp, inSamples, clev, slev, ext_leva, ext_levb, lfe_lev,
+ (FIXP_DBL)((LONG)(dmxGain5) << (METADATA_FRACT_BITS - 16)),
+ (FIXP_DBL)((LONG)(dmxGain2) << (METADATA_FRACT_BITS - 16)),
+ peak);
+
+ for (i = 0; i < 2; i++) {
+ FIXP_DBL tmp = drcComp->prevPeak[i];
+ drcComp->prevPeak[i] = peak[i];
+ peak[i] = fixMax(peak[i], tmp);
+
+ /*
+ * Convert to dBFS, apply dialnorm
+ */
+ /* descaled peak in ld64 representation */
+ FIXP_DBL ld_peak =
+ CalcLdData(peak[i]) +
+ (FIXP_DBL)((LONG)DOWNMIX_SHIFT << (DFRACT_BITS - 1 - LD_DATA_SHIFT));
+
+ /* if (peak < 1e-6) level = 1e-6f; */
+ ld_peak =
+ fMax(ld_peak, FL2FXCONST_DBL(-0.31143075889569022011284244651463f));
+
+ /* peak[i] = 20 * log(peak[i])/log(10) + 0.2f +
+ * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 20 *
+ * log(2)/log(10) * ld(peak[i]) + 0.2f +
+ * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 10 *
+ * 2*0.30102999566398119521373889472449 * ld(peak[i]) + 0.2f +
+ * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS)
+ *
+ * additional scaling with METADATA_FRACT_BITS:
+ * peak[i] = (10 * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * 64
+ * + 0.2f +
+ * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS))*2^(-METADATA_FRACT_BITS)
+ * peak[i] = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) *
+ * 2*0.30102999566398119521373889472449 * ld64(peak[i])
+ * + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i]
+ */
+ peak[i] = fMult(
+ (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)),
+ fMult(FL2FX_DBL(2 * 0.30102999566398119521373889472449f), ld_peak));
+ peak[i] +=
+ (FL2FX_DBL(0.5f) >> METADATA_INT_BITS); /* add a little bit headroom */
+ peak[i] += drcComp->smoothGain[i];
+ }
+
+ /* peak -= dialnorm + 31; */ /* this is Dolby style only */
+ peak[0] -= (FIXP_DBL)((dialnorm - drc_TargetRefLevel)
+ << (METADATA_FRACT_BITS -
+ 16)); /* peak[0] -= dialnorm - drc_TargetRefLevel */
+
+ /* peak += 11; */
+ /* this is Dolby style only */ /* RF mode output is 11dB higher */
+ /*peak += comp_TargetRefLevel - drc_TargetRefLevel;*/
+ peak[1] -=
+ (FIXP_DBL)((dialnorm - comp_TargetRefLevel)
+ << (METADATA_FRACT_BITS -
+ 16)); /* peak[1] -= dialnorm - comp_TargetRefLevel */
+
+ /* limiter gain */
+ drcComp->limGain[0] += drcComp->limDecay; /* linear limiter release */
+ drcComp->limGain[0] = fixMin(drcComp->limGain[0], -peak[0]);
+
+ drcComp->limGain[1] += 2 * drcComp->limDecay; /* linear limiter release */
+ drcComp->limGain[1] = fixMin(drcComp->limGain[1], -peak[1]);
+
+ /*************************************************************************/
+
+ /* apply limiting, return DRC gains*/
+ {
+ FIXP_DBL tmp;
+
+ tmp = drcComp->smoothGain[0];
+ if (drcComp->limGain[0] < FL2FXCONST_DBL(0.f)) {
+ tmp += drcComp->limGain[0];
+ }
+ *pDynrng = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16));
+
+ tmp = drcComp->smoothGain[1];
+ if (drcComp->limGain[1] < FL2FXCONST_DBL(0.f)) {
+ tmp += drcComp->limGain[1];
+ }
+ *pCompr = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16));
+ }
+
+ return 0;
+}
+
+DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp) {
+ return drcComp->profile[0];
+}
+
+DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp) {
+ return drcComp->profile[1];
+}